1. Field
The current invention relates to communication. More particularly, the present invention relates to a novel method and apparatus for adaptive rate selection in a wireless communication system.
2. Background
A modern communications system is required to support a variety of applications. One such communications system is a code division multiple access (CDMA) system that conforms to the “TIA/EIA/IS-95 Mobile Station-Base Station Compatibility Standard for Dual-Mode Wide-Band Spread Spectrum Cellular System,” hereinafter referred to as the IS-95 standard. The CDMA system supports voice and data communication between users over a terrestrial link. The use of CDMA techniques in a multiple access communication system is disclosed in U.S. Pat. No. 4,901,307, entitled “SPREAD SPECTRUM MULTIPLE ACCESS COMMUNICATION SYSTEM USING SATELLITE OR TERRESTRIAL REPEATERS,” and U.S. Pat. No. 5,103,459, entitled “SYSTEM AND METHOD FOR GENERATING WAVEFORMS IN A CDMA CELLULAR TELEPHONE SYSTEM,” both assigned to the assignee of the present invention and incorporated herein by reference.
In a CDMA system, communications between users are conducted through one or more base stations. In wireless communication systems, forward link refers to the channel through which signals travel from a base station to a subscriber station, and reverse link refers to channel through which signals travel from a subscriber station to a base station. By transmitting data on a reverse link to a base station, a first user on one subscriber station may communicate with a second user on a second subscriber station. The base station receives the data from the first subscriber station and routes the data to a base station serving the second subscriber station. Depending on the location of the subscriber stations, both may be served by a single base station or multiple base stations. In any case, the base station serving the second subscriber station sends the data on the forward link. Instead of communicating with a second subscriber station, a subscriber station may also communicate with a wireline telephone through a public switched telephone network (PSTN) coupled to the base station, or a terrestrial Internet through a connection with a serving base station.
Given the growing demand for wireless data applications, the need for very efficient wireless data communication systems has become increasingly significant. The IS-95 standard specifies transmitting traffic data and voice data over the forward and reverse links. A method for transmitting traffic data in code channel frames of fixed size is described in detail in U.S. Pat. No. 5,504,773, entitled “METHOD AND APPARATUS FOR THE FORMATTING OF DATA FOR TRANSMISSION,” assigned to the assignee of the present invention and incorporated by reference herein. In accordance with the IS-95 standard, the traffic data or voice data is partitioned into code channel frames that are 20 milliseconds wide with data rates as high as 14.4 Kbps.
In mobile radio communication systems, there are significant differences between the requirements for providing voice and data services (i.e., non-voice services such as Internet or fax transmissions). Unlike data services, voice services require stringent and fixed delays between speech frames. Typically, the overall one-way delay of speech frames used for transmitting voice information must be less than 100 msec. By contrast, transmission delays that occur during data (i.e., non-voice information) services can vary and larger delays then those that can be tolerated for voice services can be utilized.
Another significant difference between voice and data services is that, in contrast to data services, voice services require a fixed and common grade of service. Typically, for digital systems providing voice services, this requirement is met by using a fixed and equal transmission rate for all users and a maximum tolerable error rate for speech frames. For data services, the grade of service can vary from user to user.
Yet another difference between voice services and data services is that voice services require a reliable communication link which, in the case of a CDMA communication system, is provided using a soft handoff. A soft handoff requires the redundant transmission of the same voice information from two or more base stations to improve reliability. A soft handoff method is disclosed in U.S. Pat. No. 5,101,501, entitled “METHOD AND SYSTEM FOR PROVIDING SOFT HANDOFF IN COMMUNICATIONS IN A CDMA CELLULAR TELEPHONE SYSTEM.” This additional reliability is not required to support data services, because data packets received in error can be retransmitted.
As a mobile station moves in a mobile radio communication system, the quality of the forward link (and the capacity of the forward link to transmit data) will vary. Thus, at some moments a given forward link between a base station and a mobile station will be able to support a very high data transmission and, at other moments, the same forward link may only be able to support a much reduced data transmission rate. In order to maximize the throughput of information on the forward link, it would be desirable if the transmission of data on the forward link could be varied so as to increase the data rate during those intervals where the forward link can support a higher transmission rate.
When non-voice traffic is being sent from a base station to a mobile station on a forward link, it may be necessary to send control information from the mobile station to the base station. At times, however, even though the forward link signal may be strong, the reverse link signal may be weak, thereby resulting in a situation where the base station cannot receive control information from the mobile station. In such situations, where the forward link and the reverse link are unbalanced, it may be undesirable to increase the transmit power on the reverse link in order to improve the reception quality of the control information at the base station. For example, in CDMA systems, increasing the transmit power on the reverse link would be undesirable, as such a power increase could adversely affect the reverse link capacity seen by other mobile stations in the system. It would be desirable to have a data transmission system where the forward and reverse links associated with each mobile station were maintained in a balanced state without adversely impacting the reverse link capacity. It would be further desirable if such a system could maximize the throughput of non-voice data on individual forward links when such links are sufficiently strong to support higher data rates.
One approach to the aforementioned requirements in high data rate (HDR) systems is to keep the transmit power fixed and vary the data rate depending on the users' channel conditions. Consequently, in a modern HDR system, Access Point(s) (APs) always transmit at maximum power to only one Access Terminal (AT) in each time slot, and the AP uses rate control to adjust the maximum rate that the AT can reliably receive. An AP is a terminal allowing high data rate transmission to ATs.
As used in this document, a time slot is a time interval of finite length, e.g., 1.66 ms. A time slot can contain one or more packets. A packet is a structure, comprising a preamble, a payload, and a quality metric, e.g., a cyclical redundancy check (CRC). The preamble is used by an AT to determine whether a packet has been intended for the AT.
An exemplary HDR system defines a set of data rates, ranging from 38.4 kbps to 2.4 Mbps, at which an AP may send data packets to an AT. The data rate is selected to maintain a targeted packet error rate (PER). The AT measures the received signal to interference and noise ratio (SINR) at regular intervals, and uses the information to predict an average SINR over the next packet duration. An exemplary prediction method is disclosed in co-pending application Ser. No. 09/394,980, filed Sep. 13, 1999, entitled “SYSTEM AND METHOD FOR ACCURATELY PREDICTING SIGNAL TO INTERFERENCE AND NOISE RATIO TO IMPROVE COMMUNICATIONS SYSTEM PERFORMANCE,” now U.S. Pat. No. 6,426,971, issued to Jul. 30, 2002 to Wu et al., assigned to the assignee of the present invention and incorporated herein by reference.
FIG. 1 shows a conventional open loop rate control apparatus 100. A stream of past SINR values at instances [n−m], . . . [n−1], [n], each measured over a duration of a corresponding packet, is provided to a SINR predictor 102. The SINR predictor 102 predicts the average SINR over the next packet duration in accordance with the following equation:OL—SINRPredicted=OL—SINREstimated−K·σe  (1)
In Equation (1), OL_SINRPredicted is a SINR predicted by the open loop for the next packet, OL_SINREstimated is a SINR estimated by the open loop based on past SINR values, K is a back-off factor, and σe is a standard deviation of an error metric.
The estimated SINR may be obtained, for example, by selecting an output from a bank of low pass filters acting on past measurements of SINR. Selection of a particular filter from the filter bank may be based on an error metric, defined as a difference between the particular filter output and measured SINR over a packet duration immediately following the output. The predicted SINR is obtained by backing off from the filter output by an amount equal to the product of the back-off factor K and the standard deviation σe of the error metric. The value of the back-off factor K is determined by a back-off control loop, which ensures that a tail probability, i.e., probability that predicted SINR exceeds the measured SINR, is achieved for a certain percentage of time.
The SINRPredicted value is provided to a look up table 104 that maintains a set of SINR thresholds that represent the minimum SINR required to successfully decode a packet at each data rate. An AT (not shown) uses the look up table 104 to select the highest data rate whose SINR threshold is below the predicted SINR, and requests that an AP (not shown) send the next packet at this datarate.
The aforementioned method is an example of an open loop rate control method that determines the best rate at which to receive the next packet, based only on the measurement of the channel SINR, without any information about the decoder error rate (for packets of each data rate) at a given SINR under the prevailing channel conditions. Any open loop rate control algorithm suffers from several shortcomings, some of which are discussed below. First, a certain tail probability, e.g., 2%, does not imply a PER of 2%. This is because PER is a monotonically decreasing function of SINR, with a finite slope that depends on the coding scheme and channel conditions. However, Equation (1) assumes “brick wall” PER characteristics, i.e., a packet is guaranteed to be decoded whenever the SINR exceeds the threshold for the corresponding rate, and a packet is in error whenever the SINR falls below the threshold. Furthermore, the open loop rate control method uses a fixed set of SINR thresholds, which ensures packet error rates close to the target error rate under worst-case channel conditions. However, the performance of the decoder depends not only on the SINR, but also on channel conditions. In other words, a method that uses a fixed set of SINR thresholds for all channels achieves different packet error rates on different channels. Consequently, while the open loop method works optimally under the worst-case channel conditions, it is possible that under typical channel conditions, the method results in much lower error rates than is necessary, at the expense of diminished throughput. Additionally, a practical rate control method necessitates a small, finite set of data rates. The rate selection method always selects the nearest lower data rate in order to guarantee an acceptable PER. Thus, rate quantization results in loss of system throughput.
Therefore, there exists a need to address deficiencies of the existing method.